The term streaming media describes the playback of media on a playback device, where the media is stored on a server and continuously sent to the playback device over a network during playback. For purposes of this discussion, media and/or encoded media is defined as data of a work that includes video, audio, pictures, or another type of presentation that may be displayed, played or in some other way presented by a playback device. Typically, the playback device stores a sufficient quantity of media in a buffer at any given time during playback to prevent disruption of playback due to the playback device completing playback of all the buffered media prior to receipt of the next portion of media. Adaptive bit rate streaming or adaptive streaming involves detecting the present streaming conditions (e.g. the user's network bandwidth and CPU capacity) in real time and adjusting the quality of the streamed media accordingly. Typically, the source media is encoded at multiple bit rates and the playback device or client switches between streaming the different encodings depending on available resources.
Adaptive streaming solutions typically utilize either Hypertext Transfer Protocol (HTTP), published by the Internet Engineering Task Force and the World Wide Web Consortium as RFC 2616, or Real Time Streaming Protocol (RTSP), published by the Internet Engineering Task Force as RFC 2326, to stream media between a server and a playback device. HTTP is a stateless protocol that enables a playback device to request a byte range within a file. HTTP is described as stateless, because the server is not required to record information concerning the state of the playback device requesting information or the byte ranges requested by the playback device in order to respond to requests received from the playback device. RTSP is a network control protocol used to control streaming media servers. Playback devices issue control commands, such as “play” and “pause”, to the server streaming the media to control the playback of media files. When RTSP is utilized, the media server records the state of each client device and determines the media to stream based upon the instructions received from the client devices and the client's state.
In adaptive streaming systems, the source media is typically stored on a media server as a top level index file pointing to a number of alternate streams that contain the actual video and audio data. Each stream is typically stored in one or more container files. Different adaptive streaming solutions typically utilize different index and media containers. The Synchronized Multimedia Integration Language (SMIL) developed by the World Wide Web Consortium is utilized to create indexes in several adaptive streaming solutions including IIS Smooth Streaming developed by Microsoft Corporation of Redmond, Wash., and Flash Dynamic Streaming developed by Adobe Systems Incorporated of San Jose, Calif. HTTP Adaptive Bitrate Streaming developed by Apple Computer Incorporated of Cupertino, Calif. implements index files using an extended M3U playlist file (.M3U8), which is a text file containing a list of URIs that typically identify a media container file. The most commonly used media container formats are the MP4 container format specified in MPEG-4 Part 14 (i.e. ISO/IEC 14496-14) and the MPEG transport stream (TS) container specified in MPEG-2 Part 1 (i.e. ISO/IEC Standard 13818-1). The MP4 container format is utilized in IIS Smooth Streaming and Flash Dynamic Streaming. The TS container is used in HTTP Adaptive Bitrate Streaming.
When a playback device commences adaptive bitrate streaming, the playback device typically starts by requesting portions of media from the lowest bitrate streams (where alternative streams are available). As the playback device downloads the requested media, the playback device can measure the available bandwidth. In the event that there is additional bandwidth available, the playback device can switch to higher bitrate streams.
To start playback of the encoded media, the playback device often attempts to buffer an adequate amount of the requested encoded media be received and stored to provide a minimum amount of playback time prior to commencing the playback. Buffering encoded media can assure that there are no underflow conditions during playback. An underflow condition is when the playback device does not have the next portion of encoded media needed to continue the playback. However, the acquisition of an adequate amount of encoded media usually causes a delay in the start of playback of the encoded media by the device. This is typically not a problem when the playback is only periodically started and/or re-started.
However, recently devices such as tablets and other mobile playback devices have made it easier for users to search the encoded media and re-start the playback at various points in the playback. As such, the conventional start-up requiring an adequate amount of encoded media may cause unsatisfactory pauses or delays in presentation of the playback. As such, those skilled in the art are constantly striving to provide a playback start-up that requires less of the encoded media prior to commencement of the playback to minimize the time needed for a start and/or re-start of the playback.